OK, so I posted this in the BSD forum, but I think a few people have hinted that I should have posted in here. Sorry guys…
Anyway, here is my problem and here are all of the posts I have made thus far. If you guys can get this working for me, I’ll paypal you $50 for your help!
So, I FINALLY got Asterisk working with my Sunrocket service. Well, sort of that is…
I can make outbound calls just fine. However, inbound calls just go straight into the Sunrocket voicemail. I can’t figure out what’s going on. It’s like the Asterisk server isn’t even seeing the call come in.
I run asterisk with -rvvv and NOTHING shows up on the console when I dial in. I have my register string right, so I can’t figure out what’s going on. Here is the contents of my .conf files:
acct# = account number
ph# = phone number
sig ph# = signature phone number (second free sunrocket inbound phone number)
SIP.CONF
[general]
useragent=InnoMedia SIP MTA6328-2Re v3.0.77
register=ph#:acct#:acct#@sunrocket.com/ph#
register=sig ph#:acct#:acct#@sunrocket.com/sig ph#
bindport = 5060
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
allow=alaw
callerid = Unknown
tos=0x68
externip = 24.xxx.xxx.xxx (edited out for obvious reasons)
localnet = 192.168.0.119/255.255.255.0
context = context1 ; Send SIP callers that we don’t know about here
[sunrocket-out]
context=sipphone
type=friend
fromdomain=sunrocket.com
fromuser=ph#
username=acct#
authuser=acct#
secret=acct#
dtmfmode=inband
canreinvite=no
host=67.133.234.125
EXTENSIONS.CONF
[general]
[sipphone]
exten => _X.,1,Dial(SIP/sunrocket-out/${EXTEN})
exten => _X.,2,Hangup
[context1]
exten => _X.,1,Answer
exten => _X.,2,Wait(2)
exten => _X.,3,Playback(tt-monkeys)
exten => _X.,4,Hangup
So that’s pretty much it. I can’t figure out why inbound doesn’t work. Outbound goes great, inbound, not so great. It doesn’t even see anything.
My Asterisk box is running on Gentoo.
The Linux box is the DMZ, so it’s outside of my firewall. But, I still have specific UDP ports going to it just in case:
68, 5200, 5060-5063, 16371-16384
Also, after doing a sip show register, I get this: (I put xxxx instead of my real phone number)
asterisk*CLI> sip show registry
Host Username Refresh State
sunrocket.com:5060 xxxxxxxxxx 120 Unregistered
sunrocket.com:5060 xxxxxxxxxx 75 Registered
So, it looks like my main number is being registered, by my signature isn’t. Which isn’t a big deal, I could care less about the sig number not working…
edit
OK, if I wait about 2 mins or so, they both show up as registered:
asterisk*CLI> sip show registry
Host Username Refresh State
sunrocket.com:5060 xxxxxxxxxx 75 Registered
sunrocket.com:5060 xxxxxxxxxx 75 Registered
OK, so I did a debug, and called in, and this was the file. I xxxx out my phone numbers and stuff…
asteriskCLI> sip debug
SIP Debugging Enabled
asteriskCLI>
Sip read:
INVITE sip:678xxxxxxx@192.168.0.119 SIP/2.0
f:“Cell Phone GA” sip:678xxxxxxx@sunrocket.com;tag=3d22a68c-1dd2-11b2-8b37-b03162323164+3d26da0e
m:“Cell Phone GA” sip:678xxxxxxx@192.168.201.24:5075;transport=udp
t:sip:678xxxxxxx@sunrocket.com
i:1937664-1685040748@192.168.201.24
CSeq:1 INVITE
v:SIP/2.0/UDP 67.133.234.125:5060;branch=z9hG4bK315882abd260463023a9db0d9a62a7da-0
v:SIP/2.0/UDP 192.168.201.72:5070;branch=0
v:SIP/2.0/UDP 192.168.201.24:5060;branch=z9hG4bK38843e04bd862e91c88dd167ba900ef7.1
v:SIP/2.0/UDP 192.168.201.24:5075;branch=z9hG4bK465186393201404
Record-Route:sip:0-c0a8c94813ce19202d5d@67.133.234.125:5060;lr;transport=udp
Max-Forwards:68
Allow:INVITE,BYE,ACK,CANCEL,PRACK,REFER,OPTIONS,REGISTER,NOTIFY
c:application/sdp
k:timer
Min-SE: 1800
P-Asserted-Identity: “Cell Phone GA” sip:678xxxxxxx@sunrocket.com
l:180
v=0
o=- 323000 32300000 IN IP4 67.133.234.137
s=SIP Media Capabilities
c=IN IP4 67.133.234.137
t=0 0
m=audio 24120 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=maxptime:20
18 headers, 9 lines
Using latest request as basis request
Sending to 67.133.234.125 : 5060 (non-NAT)
Found peer 'sunrocket-out’
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 67.133.234.125:5060;branch=z9hG4bK315882abd260463023a9db0d9a62a7da-0;received=67.133.234.125;rport=5060
Via: SIP/2.0/UDP 192.168.201.72:5070;branch=0
Via: SIP/2.0/UDP 192.168.201.24:5060;branch=z9hG4bK38843e04bd862e91c88dd167ba900ef7.1
Via: SIP/2.0/UDP 192.168.201.24:5075;branch=z9hG4bK465186393201404
From: “Cell Phone GA” sip:678xxxxxxx@sunrocket.com;tag=3d22a68c-1dd2-11b2-8b37-b03162323164+3d26da0e
To: sip:678xxxxxxx@sunrocket.com;tag=as5517826b
Call-ID: 1937664-1685040748@192.168.201.24
CSeq: 1 INVITE
User-Agent: InnoMedia SIP MTA6328-2Re v3.0.77
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:678xxxxxxx@192.168.0.119
Proxy-Authenticate: Digest realm=“asterisk”, nonce="61bf1455"
Content-Length: 0
to 67.133.234.125:5060
Scheduling destruction of call ‘1937664-1685040748@192.168.201.24’ in 15000 ms
asterisk*CLI>
Sip read:
ACK sip:678xxxxxxx@192.168.0.119 SIP/2.0
f:“Cell Phone GA” sip:678xxxxxxx@sunrocket.com;tag=3d22a68c-1dd2-11b2-8b37-b03162323164+3d26da0e
m:“Cell Phone GA” sip:678xxxxxxx@192.168.201.24:5075;transport=udp
t:sip:678xxxxxxx@sunrocket.com;tag=as5517826b
i:1937664-1685040748@192.168.201.24
CSeq:1 ACK
v:SIP/2.0/UDP 67.133.234.125:5060;branch=z9hG4bK315882abd260463023a9db0d9a62a7da-0
Max-Forwards:68
Allow:INVITE,BYE,ACK,CANCEL,PRACK,REFER,OPTIONS,REGISTER,NOTIFY
k:timer
Min-SE: 1800
P-Asserted-Identity: “Cell Phone GA” sip:678xxxxxxx@sunrocket.com
l:0
13 headers, 0 lines
Destroying call '1937664-1685040748@192.168.201.24’
asterisk*CLI>
So that’s it. That’s as far as I’ve gotten as to making the inbound calls work. I have no clue what: 192.168.201.24:5075 is that keeps showing up in the debug log during an inbound call. That’s not an IP on my LAN anywhere…
Any ideas guys? $50 to whoever solves this nightmare for me. Thanks in advance.